sr-users January 2006

sr-users@lists.kamailio.org

134 participants
259 discussions

[Users] FW: [Devel] Rtpproxy coredumps :(
by Elton Machado 30 Jan '06

30 Jan '06

Hi as sugested here it goes ;) Sorry for doubling posting. Elton Machado wrote: Hi, I'm getting a problem with nathelper and rtpproxy, somethimes happens that rtpproxy core dumps, and I'm trying to understand why. Maybe I have some mistake in my nathelper configuration with the force rtp proxy. I'm using the cvs version of rtpproxy. The result of dumps is: GNU gdb 6.1.1 [FreeBSD] Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General … [View More]

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[Serusers] Sip-Rpid?
by miliczo＠sezampro.yu 30 Jan '06

30 Jan '06

Hi, In "SIP Express Router RADIUS HOWTO" by Jan Janak I've read that by adding the Sip-Rpid field in my freeradius user file "SIP Express Router can be configured to put the phone number into Remote-Party-ID header field of the SIP message" Well, my question is, can I use the Rpid field and put it in the Caller id field and how.? If anybody has done this or even thinks he has a clue, please help. Thanks

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[Users] Sending stored messages based on time
by Daniel-Constantin Mierla 30 Jan '06

30 Jan '06

Hello, I introduced the reminder functionality submitted by your patch in the development version of msilo. I changed the patch to integrate better with the module and I introduced several other functionalities. Here is the description. There is a new column "snd_time" in the table silo which specifies the time after which the message can be delivered. If the value is 0, then the message is not delivered based on timer. If the message is not successfully delivered at the specified time, … [View More]

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[Serusers] Forking based on URI. Please help.
by Leo 30 Jan '06

30 Jan '06

I am running ser 0.9.4 on FC4 with MySQL and all is working. I want to implement forking based on SIP URI. When someone dials 004888(which is a SIP URI just used for this matching), I want to fork the call to 004990 and 004100 in parallel. So I put this in my ser.cfg: if (uri==myself) { if (uri=~"^sip:004888@~*") { rewriteuri("sip:004990@sip.tcto.net:49300"); append_branch("sip:004100@sip.tcto.net:49200"); route(1); break; }; }; Route[1] { if (!t_relay()) { … [View More]

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Re: [Serusers] SER RTP QoS Monitoring
by Voipers Portugal 30 Jan '06

30 Jan '06

Well, actually, ethereal supports Real Time. There is an option when you start doing the capture that allows to show the packets in real time. Try to look at that. Jose Simoes On 1/30/06, Aisling <ashling.odriscoll(a)cit.ie> wrote: > > Hi Jose, > > > > I looked at ethereal but I don't think it's possible to use that in real > time i.e. you have to start the capture and when you think the call is > over stop the capture and examine the results. I would like … [View More]something in > real-time…that makes me think tethereal perhaps might be an option…. > > > > Thanks, > > Aisling. > > > > -----Original Message----- > *From:* Voipers Portugal [mailto:voipers@gmail.com] > *Sent:* 30 January 2006 11:47 > *To:* Aisling > *Cc:* John Todd; serusers(a)iptel.org; clona(a)cyberhouse.no > *Subject:* Re: [Serusers] SER RTP QoS Monitoring > > > > Ethereal does that automatically, so I think you can start there. As for > RTPproxy, as far as I remember, it does not support RTCP (at least it didn´t > when I tested a few months ago). > > > > Jose Simoes > > > > On 1/30/06, *Aisling* <ashling.odriscoll(a)cit.ie> wrote: > > > Hi, > > Thanks to John, Nils and Atle for the replies. So if I am to proceed > with extracting QoS info from RTCP it is now my understanding that I > would either have to have my QoS gathering application at: > > 1) the media proxy (used primarily for nat situations) > 2) where the voice is going peer to peer, it would need to be placed at > the end points. However what I'm wondering about here is would it be > possible to have a QoS collector application running on another machine > on the same network as the two endpoints (obviously presuming they're > all on the same network) which could intercept the sender and receiver > reports? And if so does anyone have any idea if there's another way to > distinguish RTCP packets without knowing the IP address or port? > > Many thanks, > Aisling. > > -----Original Message----- > From: John Todd [mailto: jtodd(a)loligo.com] > Sent: 29 January 2006 22:33 > To: Aisling O'Driscoll; serusers(a)iptel.org > Subject: Re: [Serusers] SER RTP QoS Monitoring > > Hello - > Your method #1 (RTCP) is the best solution, it would seem. Method > #2 (INFO) is sort-of supported in some devices, like the Cisco 79xx > series with SIP software - they support insertion of some basic call > element data into the BYE message. Method #3 is complex, as it would > require the SNMP collector to have full knowledge of all call states > on the devices that were queried (though possibly this is not more > complex than the RTCP collection process, either.) In any of these > three cases, SER does not collect the call quality data - the > endpoints or media proxies are the only places that see this data. > If you run a mediaproxy in conjunction with SER, then it may be > possible to collect and report on that data, but there are drawbacks > to a design of that nature. > > For the long term, I would suggest taking a look at this draft, > which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to > be transmitted via SIP messages between media-handling entities and > other reporting systems: > > > http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary- > 08.txt > > While we are still a while away from seeing vendor (or open source) > implementations, this seems like a reasonable proposal. It allows > for SIP-based publication or subscription of call quality events by > various network elements, and it will permit arbitrary querying of > this data at any time interval (not just at the end of the call.) > > JT > > > > >Hi, > > > >I want to be able to extract QoS parameters such as jitter, delay and > >packet loss from VoIP calls made using my SER(both peer to peer for > >my public clients and also for clients where the voice will be routed > >with mediaproxy). I would like to develop something as opposed to > >purchasing a software package. > > > >>From searching the Internet the last few days I have seen that the > >following options seem to be available: > > > >1) RTCP > >RTCP stream will contain this information. The problem is that > >sniffers cannot clearly distinguish RTCP packets from other UDP > >packets unless something like an IP address or specific UDP port is > >supplied. I had considered using TCPDUMP but then where do I position > >this in the network when voice is going peer to peer between client > >and I'm not sure how this can work to monitor all calls when I might > >not be aware of the end user IP addresses. > > > >2) SIP INFO extension > >This also seem to be an option but most off the shelf phones don't > >seem to support this and this would required modification of a SIP ua. > > > >3) SNMP > >I thought maybe SNMP might be an option but the SER snmp module no > >longer exists... > > > >Does anyone have any comments on the above? Are the statements that I > >made correct or can anyone think of other ways to monitor the voice > >QoS? I am trying to understand how commercial applications have > >accomplished this. > > > >Many thanks in advance for advice. > >Aisling. > > > >>---- Original Message ---- > >>From: clona(a)cyberhouse.no > >>To: ashling.odriscoll(a)cit.ie > >>Subject: Re: [Serusers] Status of SER SNMP module > >>Date: Sun, 29 Jan 2006 20:30:02 +0100 > >> > >>> > >>>Hi Aisling, > >>> > >>>The SNMP module is dead as a killed turtle :( > >>> > >>>-Atle > >>>* Aisling O'Driscoll < ashling.odriscoll(a)cit.ie> [060129 20:26]: > >>>> Hello, > >>>> > >>>> I'm just wondering what the current status of the SER SNMP module > >>is? > >>>> Is it currently supported and is there somewhere I can find > >>>> documentation on it? > >>>> > >>>> Many thanks, > > >>> Aisling. > > > >_______________________________________________ > >Serusers mailing list > >Serusers(a)iptel.org > >http://mail.iptel.org/mailman/listinfo/serusers > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended only > for the person to whom it is addressed. Its contents may be protected by > legal and/or professional privilege. Should it be received by you in > error please contact the sender at the above quoted email address. Any > unauthorised form of reproduction of this message is strictly > prohibited. The Institute does not guarantee the security of any > information electronically transmitted and is not liable if the > information contained in this communication is not a proper and complete > record of the message as transmitted by the sender nor for any delay in > its receipt. > > > -------------------Legal Disclaimer--------------------------------------- > > > The above electronic mail transmission is confidential and intended only > for the person to whom it is addressed. Its contents may be protected by > legal and/or professional privilege. Should it be received by you in error > please contact the sender at the above quoted email address. Any > unauthorised form of reproduction of this message is strictly prohibited. > The Institute does not guarantee the security of any information > electronically transmitted and is not liable if the information contained in > this communication is not a proper and complete record of the message as > transmitted by the sender nor for any delay in its receipt. > > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > > > -------------------Legal Disclaimer--------------------------------------- > The above electronic mail transmission is confidential and intended only for > the person to whom it is addressed. Its contents may be protected by legal > and/or professional privilege. Should it be received by you in error please > contact the sender at the above quoted email address. Any unauthorised form > of reproduction of this message is strictly prohibited. The Institute does > not guarantee the security of any information electronically transmitted and > is not liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the sender nor > for any delay in its receipt. > -------------------Legal Disclaimer--------------------------------------- > The above electronic mail transmission is confidential and intended only for > the person to whom it is addressed. Its contents may be protected by legal > and/or professional privilege. Should it be received by you in error please > contact the sender at the above quoted email address. Any unauthorised form > of reproduction of this message is strictly prohibited. The Institute does > not guarantee the security of any information electronically transmitted and > is not liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the sender nor > for any delay in its receipt. > [View Less]

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FW: [Serusers] SER RTP QoS Monitoring
by Aisling 30 Jan '06

30 Jan '06

Hi Jose, I looked at ethereal but I dont think its possible to use that in real time i.e. you have to start the capture and when you think the call is over stop the capture and examine the results. I would like something in real-timethat makes me think tethereal perhaps might be an option. Thanks, Aisling. -----Original Message----- From: Voipers Portugal [mailto:voipers@gmail.com] Sent: 30 January 2006 11:47 To: Aisling Cc: John Todd; serusers(a)lists.iptel.org; clona(a)cyberhouse.… [View More]no Subject: Re: [Serusers] SER RTP QoS Monitoring Ethereal does that automatically, so I think you can start there. As for RTPproxy, as far as I remember, it does not support RTCP (at least it didn´t when I tested a few months ago). Jose Simoes On 1/30/06, Aisling <ashling.odriscoll(a)cit.ie> wrote: Hi, Thanks to John, Nils and Atle for the replies. So if I am to proceed with extracting QoS info from RTCP it is now my understanding that I would either have to have my QoS gathering application at: 1) the media proxy (used primarily for nat situations) 2) where the voice is going peer to peer, it would need to be placed at the end points. However what I'm wondering about here is would it be possible to have a QoS collector application running on another machine on the same network as the two endpoints (obviously presuming they're all on the same network) which could intercept the sender and receiver reports? And if so does anyone have any idea if there's another way to distinguish RTCP packets without knowing the IP address or port? Many thanks, Aisling. -----Original Message----- From: John Todd [mailto: jtodd(a)loligo.com] Sent: 29 January 2006 22:33 To: Aisling O'Driscoll; serusers(a)lists.iptel.org Subject: Re: [Serusers] SER RTP QoS Monitoring Hello - Your method #1 (RTCP) is the best solution, it would seem. Method #2 (INFO) is sort-of supported in some devices, like the Cisco 79xx series with SIP software - they support insertion of some basic call element data into the BYE message. Method #3 is complex, as it would require the SNMP collector to have full knowledge of all call states on the devices that were queried (though possibly this is not more complex than the RTCP collection process, either.) In any of these three cases, SER does not collect the call quality data - the endpoints or media proxies are the only places that see this data. If you run a mediaproxy in conjunction with SER, then it may be possible to collect and report on that data, but there are drawbacks to a design of that nature. For the long term, I would suggest taking a look at this draft, which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to be transmitted via SIP messages between media-handling entities and other reporting systems: http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary- 08.txt While we are still a while away from seeing vendor (or open source) implementations, this seems like a reasonable proposal. It allows for SIP-based publication or subscription of call quality events by various network elements, and it will permit arbitrary querying of this data at any time interval (not just at the end of the call.) JT >Hi, > >I want to be able to extract QoS parameters such as jitter, delay and >packet loss from VoIP calls made using my SER(both peer to peer for >my public clients and also for clients where the voice will be routed >with mediaproxy). I would like to develop something as opposed to >purchasing a software package. > >>From searching the Internet the last few days I have seen that the >following options seem to be available: > >1) RTCP >RTCP stream will contain this information. The problem is that >sniffers cannot clearly distinguish RTCP packets from other UDP >packets unless something like an IP address or specific UDP port is >supplied. I had considered using TCPDUMP but then where do I position >this in the network when voice is going peer to peer between client >and I'm not sure how this can work to monitor all calls when I might >not be aware of the end user IP addresses. > >2) SIP INFO extension >This also seem to be an option but most off the shelf phones don't >seem to support this and this would required modification of a SIP ua. > >3) SNMP >I thought maybe SNMP might be an option but the SER snmp module no >longer exists... > >Does anyone have any comments on the above? Are the statements that I >made correct or can anyone think of other ways to monitor the voice >QoS? I am trying to understand how commercial applications have >accomplished this. > >Many thanks in advance for advice. >Aisling. > >>---- Original Message ---- >>From: clona(a)cyberhouse.no >>To: ashling.odriscoll(a)cit.ie >>Subject: Re: [Serusers] Status of SER SNMP module >>Date: Sun, 29 Jan 2006 20:30:02 +0100 >> >>> >>>Hi Aisling, >>> >>>The SNMP module is dead as a killed turtle :( >>> >>>-Atle >>>* Aisling O'Driscoll < <mailto:ashling.odriscoll@cit.ie> ashling.odriscoll(a)cit.ie> [060129 20:26]: >>>> Hello, >>>> >>>> I'm just wondering what the current status of the SER SNMP module >>is? >>>> Is it currently supported and is there somewhere I can find >>>> documentation on it? >>>> >>>> Many thanks, > >>> Aisling. > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. [View Less]

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[Serusers] Whiskey & Beer for SNMP module
by Thomas Nilsen 30 Jan '06

30 Jan '06

Hello! Two cases of beer or two bottles of whiskey to first developer that release a simple snmp module that provide equal status equal to the info accessible through the fifo. Deadline for first sneak preview is next Friday! :-) - Cheers Thomas

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[Serusers] failover again
by hgaillac-sip＠yahoo.fr 30 Jan '06

30 Jan '06

Hi all, I wish to set ser (outbound proxy) with failover support : SER1 sip agents ======= ======= Asterisk Farm ==sip/pstn gateway SER2 SER1 and SER2 use SQL DB if one of the SERs is down how can we get the other available for the sip agents for example we need to reboot a SER when the ser.cfg has changed . I look at dispatcher module ! Regards Harry ___________________________________________________________________________ … [View More]

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[Serusers] Ser server
by Ofer Zaarur 29 Jan '06

29 Jan '06

Hi All, I'm trying to connect X-lite to ser server that installed as local server. The test configuration is running in local network and the IP is fix. please let me know if you have any idea what is going on. thanks, Ofer log from the Xlite © 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1105d build stamp 99999 License key: 3FA27C9F993AC5240ADFBC4466EBEEA5 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 … [View More]Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5061 Established SIP protocol listen on: 192.168.1.45:5061 Discovered Single Mapped Port Symmetric NAT Firewall SIP: 192.168.1.45:5061 RTP: 192.168.1.45:8000 NAT: 151.203.64.98 Discovering external SIP port on symmetric firewall... Discovered external SIP port on symmetric firewall: 50402 RECEIVE TIME: 405492948 RECEIVE << 64.69.76.23:3478 PROXY#0: 192.168.1.45:5060 SEND TIME: 405493076 SEND >> 192.168.1.45:5060 REGISTER sip:192.168.1.45 SIP/2.0 v: SIP/2.0/UDP 192.168.1.45:5061;rport;branch=z9hG4bK11776FC28025ABB2C29C1BD89E88795E f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45> m: "1234" <sip:1234@192.168.1.45:5061> i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5434 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105d l: 0 RECEIVE TIME: 405493262 RECEIVE << 192.168.1.45:5060 SIP/2.0 483 Too Many Hops v: SIP/2.0/UDP 192.168.1.45:5061;rport=5061;branch=z9hG4bK11776FC28025ABB2C29C1BD89E88795E f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45>;tag=b27e1a1d33761e85846fc98f5f3a7e58.bee7 i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5434 REGISTER Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.45:5060 "Noisy feedback tells: pid=12541 req_src_ip=192.168.1.45 req_src_port=5060 in_uri=sip:192.168.1.45 out_uri=sip:192.168.1.45 via_cnt==71" SEND TIME: 405513508 SEND >> 192.168.1.45:5060 REGISTER sip:192.168.1.45 SIP/2.0 v: SIP/2.0/UDP 192.168.1.45:5061;rport;branch=z9hG4bK705C4BCDC57F933D959A1C991F8A4DB0 f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45> m: "1234" <sip:1234@192.168.1.45:5061> i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5435 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105d l: 0 RECEIVE TIME: 405513686 RECEIVE << 192.168.1.45:5060 SIP/2.0 483 Too Many Hops v: SIP/2.0/UDP 192.168.1.45:5061;rport=5061;branch=z9hG4bK705C4BCDC57F933D959A1C991F8A4DB0 f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6822 i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5435 REGISTER Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.45:5060 "Noisy feedback tells: pid=12541 req_src_ip=192.168.1.45 req_src_port=5060 in_uri=sip:192.168.1.45 out_uri=sip:192.168.1.45 via_cnt==71" [View Less]

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[Serusers] rtpproxy
by Toygun Mavinil 29 Jan '06

29 Jan '06

Hi there, I try to use ser 0.9.6 on fc4 with mysql rtpproxy because of nat No errors starting ser On local, no problen betweeb ip phones and gateways but one way voice problen on sjphone On wan, always voice goes one way, caller can nor hear but callee can hear and can not talk FC4 gets wan ip directly, all firewalls are off. I tried with mediaproxy module, nothing changed, Here is ser.cfg (from configs.GettingStarted); # $Id: nat-rtpproxy.cfg 9 2005-08-19 debug=3 fork=yes log_stderror=no … [View More]listen=wan ip port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("FQDN","subscriber")) { proxy_challenge("FQDN","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ------------ Toygun [View Less]

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sr-users January 2006